Chapter6 PDF

Fundamentals of Multimedia, Chapter 6
Chapter 6
Basics of Digital Audio
6.1 Digitization of Sound

6.2 MIDI: Musical Instrument Digital Interface
6.3 Quantization and Transmission of Audio
6.4 Further Exploration
c Prentice Hall 2003

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6.1 Digitization of Sound

What is Sound?
Sound is a wave phenomenon like light, but is macroscopic
and involves molecules of air being compressed and expanded
under the action of some physical device.
(a) For example, a speaker in an audio system vibrates back
and forth and produces a longitudinal pressure wave that
we perceive as sound.
(b) Since sound is a pressure wave, it takes on continuous
values, as opposed to digitized ones.

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(c) Even though such pressure waves are longitudinal, they

still have ordinary wave properties and behaviors, such as
reflection (bouncing), refraction (change of angle when
entering a medium with a different density) and diffraction
(bending around an obstacle).
(d) If we wish to use a digital version of sound waves we must
form digitized representations of audio information.
Link to physical description of sound waves.

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Digitization
Digitization means conversion to a stream of numbers, and
preferably these numbers should be integers for efficiency.
Fig. 6.1 shows the 1-dimensional nature of sound: amplitude
values depend on a 1D variable, time. (And note that images
depend instead on a 2D set of variables, x and y).

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Amplitude
Time
Fig. 6.1:
An analog signal: continuous measurement of
pressure wave.

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The graph in Fig. 6.1 has to be made digital in both time

and amplitude. To digitize, the signal must be sampled in
each dimension: in time, and in amplitude.
(a) Sampling means measuring the quantity we are interested in, usually
at evenly-spaced intervals.
(b) The first kind of sampling, using measurements only at evenly spaced
time intervals, is simply called, sampling. The rate at which it is
performed is called the sampling frequency (see Fig. 6.2(a)).
(c) For audio, typical sampling rates are from 8 kHz (8,000 samples per
second) to 48 kHz. This range is determined by Nyquist theorem
discussed later.
(d) Sampling in the amplitude or voltage dimension is called quantization. Fig. 6.2(b) shows this kind of sampling.

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Amplitude
Amplitude
Time
Time
(a)
Fig. 6.2:
(b)
Sampling and Quantization. (a): Sampling the analog signal
in the time dimension. (b): Quantization is sampling the analog signal

in the amplitude dimension.

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Thus to decide how to digitize audio data we need to answer

the following questions:
1. What is the sampling rate?

2. How finely is the data to be quantized, and is quantization
uniform?
3. How is audio data formatted? (file format)

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Nyquist Theorem
Signals can be decomposed into a sum of sinusoids. Fig. 6.3
shows how weighted sinusoids can build up quite a complex
signal.

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Fundamental
frequency
+ 0.5
2 fundamental
+ 0.33
3 fundamental
+ 0.25
4 fundamental
+ 0.5
5 fundamental
Fig. 6.3: Building up a complex signal by superposing sinusoids
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Whereas frequency is an absolute measure, pitch is generally relative a perceptual subjective quality of sound.
(a) Pitch and frequency are linked by setting the note A above
middle C to exactly 440 Hz.
(b) An octave above that note takes us to another A note.
An octave corresponds to doubling the frequency. Thus
with the middle A on a piano (A4 or A440) set
to 440 Hz, the next A up is at 880 Hz, or one octave
above.
(c) Harmonics: any series of musical tones whose frequencies
are integral multiples of the frequency of a fundamental
tone: Fig. 6.
(d) If we allow non-integer multiples of the base frequency, we
allow non-A notes and have a more complex resulting
sound.
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The Nyquist theorem states how frequently we must sample

in time to be able to recover the original sound.
(a) Fig. 6.4(a) shows a single sinusoid: it is a single, pure, frequency
(only electronic instruments can create such sounds).
(b) If sampling rate just equals the actual frequency, Fig. 6.4(b) shows
that a false signal is detected: it is simply a constant, with zero
frequency.
(c) Now if sample at 1.5 times the actual frequency, Fig. 6.4(c) shows
that we obtain an incorrect (alias) frequency that is lower than the
correct one it is half the correct one (the wavelength, from peak
to peak, is double that of the actual signal).
(d) Thus for correct sampling we must use a sampling rate equal to at
least twice the maximum frequency content in the signal. This rate
is called the Nyquist rate.
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(a)
(b)
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(c)
Fig. 6.4: Aliasing. (a): A single frequency. (b): Sampling at
exactly the frequency produces a constant. (c): Sampling at
1.5 times per cycle produces an alias perceived frequency.
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Nyquist Theorem: If a signal is band-limited, i.e., there is

a lower limit f1 and an upper limit f2 of frequency components in the signal, then the sampling rate should be
at least 2(f2 f1).
Nyquist frequency: half of the Nyquist rate.
Since it would be impossible to recover frequencies higher than Nyquist
frequency in any event, most systems have an antialiasing filter that
restricts the frequency content in the input to the sampler to a range
at or below Nyquist frequency.
The relationship among the Sampling Frequency, True Frequency, and the Alias Frequency is as follows:
falias = fsampling ftrue ,
for
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ftrue < fsampling < 2 ftrue
(6.1)

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Apparent frequency (kHz)
In general, the apparent frequency of a sinusoid is the lowest

frequency of a sinusoid that has exactly the same samples
as the input sinusoid. Fig. 6.5 shows the relationship of the
apparent frequency to the input frequency.
4
3
2
1
0
0
4
6
8
True frequency (kHz)
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Fig. 6.5: Folding of sinusoid frequency which is sampled at

8,000 Hz. The folding frequency, shown dashed, is 4,000 Hz.
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Signal to Noise Ratio (SNR)

The ratio of the power of the correct signal and the noise is
called the signal to noise ratio (SNR) a measure of the
quality of the signal.
The SNR is usually measured in decibels (dB), where 1 dB is
a tenth of a bel. The SNR value, in units of dB, is defined in
terms of base-10 logarithms of squared voltages, as follows:
SN R = 10 log10
2
Vsignal
2
Vnoise
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= 20 log10
Vsignal
Vnoise
(6.2)

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a) The power in a signal is proportional to the square of the

voltage. For example, if the signal voltage Vsignal is 10
times the noise, then the SNR is 20 log10(10) = 20dB.
b) In terms of power, if the power from ten violins is ten
times that from one violin playing, then the ratio of power
is 10dB, or 1B.
c) To know:
Power 10; Signal Voltage 20.
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The usual levels of sound we hear around us are described

in terms of decibels, as a ratio to the quietest sound we are
capable of hearing. Table 6.1 shows approximate levels for
these sounds.
Table 6.1: Magnitude levels of common sounds, in decibels
Threshold of hearing
Rustle of leaves
Very quiet room
Average room
Conversation
Busy street
Loud radio
Train through station
Riveter
Threshold of discomfort
Threshold of pain
Damage to ear drum
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0
10
20
40
60
70
80
90
100
120
140
160

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Signal to Quantization Noise Ratio (SQNR)

Aside from any noise that may have been present in the
original analog signal, there is also an additional error that
results from quantization.
(a) If voltages are actually in 0 to 1 but we have only 8 bits
in which to store values, then effectively we force all continuous values of voltage into only 256 different values.
(b) This introduces a roundoff error. It is not really noise.
Nevertheless it is called quantization noise (or quantization error).
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The quality of the quantization is characterized by the Signal

to Quantization Noise Ratio (SQNR).
(a) Quantization noise: the difference between the actual
value of the analog signal, for the particular sampling time,
and the nearest quantization interval value.
(b) At most, this error can be as much as half of the interval.
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(c) For a quantization accuracy of N bits per sample, the

SQNR can be simply expressed:
SQN R = 20 log10 V
Vsignal
quan noise
N 1
1
2
= 20 log10 2
= 20 N log 2 = 6.02 N (dB)
(6.3)
Notes:
(a) We map the maximum signal to 2N 1 1 (' 2N 1) and
the most negative signal to 2N 1 .
(b) Eq. (6.3) is the Peak signal-to-noise ratio, PSQNR: peak
signal and peak noise.
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(c) The dynamic range is the ratio of maximum to minimum

absolute values of the signal: Vmax/Vmin. The max abs.
value Vmax gets mapped to 2N 1 1; the min abs. value
Vmin gets mapped to 1. Vmin is the smallest positive
voltage that is not masked by noise. The most negative
signal, Vmax, is mapped to 2N 1 .
(d) The quantization interval is V =(2Vmax)/2N , since there
are 2N intervals. The whole range Vmax down to (Vmax
V /2) is mapped to 2N 1 1.
(e) The maximum noise, in terms of actual voltages, is half
the quantization interval: V /2=Vmax/2N .
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6.02N is the worst case. If the input signal is sinusoidal,

the quantization error is statistically independent, and its
magnitude is uniformly distributed between 0 and half of the
interval, then it can be shown that the expression for the
SQNR becomes:
SQN R = 6.02N +1.76(dB)
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(6.4)

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Linear and Non-linear Quantization

Linear format: samples are typically stored as uniformly
quantized values.
Non-uniform quantization: set up more finely-spaced levels
where humans hear with the most acuity.
Webers Law stated formally says that equally perceived
differences have values proportional to absolute levels:
Response Stimulus/Stimulus
(6.5)
Inserting a constant of proportionality k, we have a differential equation that states:

dr = k (1/s) ds
(6.6)
with response r and stimulus s.

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Integrating, we arrive at a solution

r = k ln s + C
with constant of integration C.
Stated differently, the solution is
r = k ln(s/s0)
(6.7)
(6.8)
s0 = the lowest level of stimulus that causes a response

(r = 0 when s = s0).
Nonlinear quantization works by first transforming an analog
signal from the raw s space into the theoretical r space, and
then uniformly quantizing the resulting values.
Such a law for audio is called -law encoding, (or u-law). A
very similar rule, called A-law, is used in telephony in Europe.
The equations for these very similar encodings are as follows:
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-law:
sgn(s)
ln
r =
ln(1 + )
A-law:
)
s

1 + ,
sp

A
s ,
s
p
1+lnA
r =

s

1
sp
(6.9)

s
1
sp
A

sgn(s) 1 + lnA s , 1 s 1
sp
A
sp
1+lnA
(
where sgn(s) =
(6.10)
1
if s > 0,
1 otherwise
Fig. 6.6 shows these curves. The parameter is set to

= 100 or = 255; the parameter A for the A-law encoder
is usually set to A = 87.6.
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-law or A-law
0.8
0.6
-law: = 100
r : -law or A-law
0.4
A-law: A = 87.6
0.2
0
0.2
0.4
0.6
0.8
1
1
0.8 0.6 0.4 0.2
0
s/sp
0.2
0.4
0.6
0.8
Fig. 6.6: Nonlinear transform for audio signals

The -law in audio is used to develop a nonuniform quantization rule for sound: uniform quantization of r gives finer
resolution in s at the quiet end.
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Audio Filtering
Prior to sampling and AD conversion, the audio signal is
also usually filtered to remove unwanted frequencies. The
frequencies kept depend on the application:
(a) For speech, typically from 50Hz to 10kHz is retained, and other frequencies are blocked by the use of a band-pass filter that screens
out lower and higher frequencies.
(b) An audio music signal will typically contain from about 20Hz up to
20kHz.
(c) At the DA converter end, high frequencies may reappear in the output because of sampling and then quantization, smooth input
signal is replaced by a series of step functions containing all possible
frequencies.
(d) So at the decoder side, a lowpass filter is used after the DA circuit.
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Audio Quality vs. Data Rate

The uncompressed data rate increases as more bits are used
for quantization. Stereo: double the bandwidth. to transmit
a digital audio signal.
Table 6.2: Data rate and bandwidth in sample audio applications
Quality
Sample
Bits per
Mono/
Data Rate
Frequency
Rate
Sample
Stereo
(uncompressed)
Band
(kB/sec)
(KHz)
(KHz)
Telephone
Mono
0.200-3.4
AM Radio
11.025
Mono
11.0
0.1-5.5
FM Radio
22.05
16
Stereo
88.2
0.02-11
44.1
16
Stereo
176.4
0.005-20
48
16
Stereo
192.0
0.005-20
192 (max)
24 (max)
6 channels
1,200.0 (max)
0-96 (max)
CD
DAT
DVD Audio
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Synthetic Sounds
1. FM (Frequency Modulation): one approach to generating
synthetic sound:
x(t) = A(t) cos[ct + I(t) cos(m t+m)+c]
(6.11)
Link to details.
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cos (4 t)
1.0
1.0
0.5
0.5
Magnitude
Magnitude
cos (2 t)
0.0
0.5
1.0
0.0
0.2
0.4 0.6
Time
0.8
0.0
0.5
1.0
1.0
0.0
0.2
(a)
0.5
0.5
Magnitude
Magnitude
1.0
0.0
0.5
0.2
0.4 0.6
Time
1.0
cos (2 t + cos (4 t))
1.0
0.0
0.8
(b)
cos (cos (4 t))
1.0
0.4 0.6
Time
0.8
1.0
(c)
0.0
0.5
1.0
0.0
0.2
0.4 0.6
Time
0.8
1.0
(d)
Fig. 6.7: Frequency Modulation. (a): A single frequency. (b): Twice the
frequency. (c): Usually, FM is carried out using a sinusoid argument to a
sinusoid. (d): A more complex form arises from a carrier frequency, 2t
and a modulating frequency 4t cosine inside the sinusoid.
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2. Wave Table synthesis: A more accurate way of generating

sounds from digital signals. Also known, simply, as sampling.
In this technique, the actual digital samples of sounds from
real instruments are stored. Since wave tables are stored in
memory on the sound card, they can be manipulated by software so that sounds can be combined, edited, and enhanced.
Link to details.
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6.2 MIDI: Musical Instrument Digital Interface

Use the sound cards defaults for sounds: use a simple
scripting language and hardware setup called MIDI.
MIDI Overview
(a) MIDI is a scripting language it codes events that stand for the
production of sounds. E.g., a MIDI event might include values for
the pitch of a single note, its duration, and its volume.
(b) MIDI is a standard adopted by the electronic music industry for controlling devices, such as synthesizers and sound cards, that produce
music.
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(c) The MIDI standard is supported by most synthesizers, so sounds

created on one synthesizer can be played and manipulated on another
synthesizer and sound reasonably close.
(d) Computers must have a special MIDI interface, but this is incorporated into most sound cards. The sound card must also have both
D/A and A/D converters.
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MIDI Concepts
MIDI channels are used to separate messages.
(a) There are 16 channels numbered from 0 to 15. The channel forms
the last 4 bits (the least significant bits) of the message.
(b) Usually a channel is associated with a particular instrument: e.g.,
channel 1 is the piano, channel 10 is the drums, etc.
(c) Nevertheless, one can switch instruments midstream, if desired, and
associate another instrument with any channel.
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System messages
(a) Several other types of messages, e.g. a general message for all instruments indicating a change in tuning or timing.
(b) If the first 4 bits are all 1s, then the message is interpreted as a
system common message.
The way a synthetic musical instrument responds to a MIDI

message is usually by simply ignoring any play sound message that is not for its channel.
If several messages are for its channel, then the instrument responds,
provided it is multi-voice, i.e., can play more than a single note at
once.
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It is easy to confuse the term voice with the term timbre

the latter is MIDI terminology for just what instrument that
is trying to be emulated, e.g. a piano as opposed to a violin:
it is the quality of the sound.
(a) An instrument (or sound card) that is multi-timbral is one that is
capable of playing many different sounds at the same time, e.g.,
piano, brass, drums, etc.
(b) On the other hand, the term voice, while sometimes used by musicians to mean the same thing as timbre, is used in MIDI to mean
every different timbre and pitch that the tone module can produce at
the same time.
Different timbres are produced digitally by using a patch

the set of control settings that define a particular timbre.
Patches are often organized into databases, called banks.
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General MIDI: A standard mapping specifying what instruments (what patches) will be associated with what channels.
(a) In General MIDI, channel 10 is reserved for percussion instruments,
and there are 128 patches associated with standard instruments.
(b) For most instruments, a typical message might be a Note On message
(meaning, e.g., a keypress and release), consisting of what channel,
what pitch, and what velocity (i.e., volume).
(c) For percussion instruments, however, the pitch data means which
kind of drum.
(d) A Note On message consists of status byte which channel,
what pitch followed by two data bytes. It is followed by a Note
Off message, which also has a pitch (which note to turn off) and a
velocity (often set to zero).
Link to General MIDI Instrument Patch Map
Link to General MIDI Percussion Key Map

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The data in a MIDI status byte is between 128 and 255; each
of the data bytes is between 0 and 127. Actual MIDI bytes
are 10-bit, including a 0 start and 0 stop bit.
Stop One Start

bit byte bit
Transmitting
device
0 11101110 0 0 10110011 0 0 110111010
Synthesizer
Fig. 6.8: Stream of 10-bit bytes; for typical MIDI messages,

these consist of {Status byte, Data Byte, Data Byte} =
{Note On, Note Number, Note Velocity}
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A MIDI device often is capable of programmability, and also

can change the envelope describing how the amplitude of a
sound changes over time.
Fig. 6.9 shows a model of the response of a digital instrument
to a Note On message:
Attack
Amplitude
Decay
Note off
Sustain
Release
t
Note on
Fig. 6.9: Stages of amplitude versus time for a music note

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Hardware Aspects of MIDI

The MIDI hardware setup consists of a 31.25 kbps serial connection. Usually, MIDI-capable units are either Input devices
or Output devices, not both.
A traditional synthesizer is shown in Fig. 6.10:
Keyboard
Pitch bend
wheel
Modulation
wheel
Fig. 6.10: A MIDI synthesizer

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The physical MIDI ports consist of 5-pin connectors for IN

and OUT, as well as a third connector called THRU.
(a) MIDI communication is half-duplex.
(b) MIDI IN is the connector via which the device receives all
MIDI data.
(c) MIDI OUT is the connector through which the device
transmits all the MIDI data it generates itself.
(d) MIDI THRU is the connector by which the device echoes
the data it receives from MIDI IN. Note that it is only the
MIDI IN data that is echoed by MIDI THRU all the
data generated by the device itself is sent via MIDI OUT.
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A typical MIDI sequencer setup is shown in Fig. 6.11:
OUT
IN
THRU
Master keyboard
THRU
IN
MIDI module A
IN
THRU
MIDI module B
etc.
Fig. 6.11: A typical MIDI setup

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Structure of MIDI Messages

MIDI messages can be classified into two types: channel
messages and system messages, as in Fig. 6.12:
Voice messages
Channel messages
Mode messages
MIDI messages
Common messages
System messages
Real-time messages
Exclusive messages
Fig. 6.12: MIDI message taxonomy
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A. Channel messages: can have up to 3 bytes:

a) The first byte is the status byte (the opcode, as it were); has its most
significant bit set to 1.
b) The 4 low-order bits identify which channel this message belongs to
(for 16 possible channels).
c) The 3 remaining bits hold the message. For a data byte, the most
significant bit is set to 0.
A.1. Voice messages:

a) This type of channel message controls a voice, i.e., sends information
specifying which note to play or to turn off, and encodes key pressure.
b) Voice messages are also used to specify controller effects such as
sustain, vibrato, tremolo, and the pitch wheel.
c) Table 6.3 lists these operations.
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Table 6.3: MIDI voice messages.

Voice Message
Status Byte
Data Byte1
Data Byte2
Note Off
&H8n
Key number
Note Off velocity
Note On
&H9n
Key number
Note On velocity
Poly. Key Pressure
&HAn
Key number
Amount
Control Change
&HBn
Controller num.
Controller value
Program Change
&HCn
Program number
None
Channel Pressure
&HDn
Pressure value
None
Pitch Bend
&HEn
MSB
LSB
(** &H indicates hexadecimal, and n in the status byte hex value stands
for a channel number. All values are in 0..127 except Controller number,
which is in 0..120)
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A.2. Channel mode messages:

a) Channel mode messages: special case of the Control Change message
opcode B (the message is &HBn, or 1011nnnn).
b) However, a Channel Mode message has its first data byte in 121
through 127 (&H797F).
c) Channel mode messages determine how an instrument processes MIDI
voice messages: respond to all messages, respond just to the correct
channel, dont respond at all, or go over to local control of the instrument.
d) The data bytes have meanings as shown in Table 6.4.
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Table 6.4: MIDI Mode Messages

1st Data Byte
Description
Meaning of 2nd
Data Byte
&H79
Reset all controllers
None; set to 0
&H7A
Local control
0 = off; 127 = on
&H7B
All notes off
None; set to 0
&H7C
Omni mode off
None; set to 0
&H7D
Omni mode on
None; set to 0
&H7E
Mono mode on (Poly mode off)
Controller number
&H7F
Poly mode on (Mono mode off)
None; set to 0
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B. System Messages:
a) System messages have no channel number commands

that are not channel specific, such as timing signals for
synchronization, positioning information in pre-recorded
MIDI sequences, and detailed setup information for the
destination device.
b) Opcodes for all system messages start with &HF.
c) System messages are divided into three classifications, according to their use:
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B.1. System common messages: relate to timing or positioning.

Table 6.5: MIDI System Common messages.
System Common Message
Status Byte
Number of Data Bytes
MIDI Timing Code
&HF1
Song Position Pointer
&HF2
Song Select
&HF3
Tune Request
&HF6
None
EOX (terminator)
&HF7
None
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B.2. System real-time messages: related to synchronization.
Table 6.6: MIDI System Real-Time messages.
System Real-Time Message Status Byte

Timing Clock
&HF8
Start Sequence
&HFA
Continue Sequence
&HFB
Stop Sequence
&HFC
Active Sensing
&HFE
System Reset
&HFF
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B.3. System exclusive message: included so that the MIDI

standard can be extended by manufacturers.
a) After the initial code, a stream of any specific messages
can be inserted that apply to their own product.
b) A System Exclusive message is supposed to be terminated by a terminator byte &HF7, as specified in Table 6.
c) The terminator is optional and the data stream may
simply be ended by sending the status byte of the next
message.
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General MIDI
General MIDI is a scheme for standardizing the assignment
of instruments to patch numbers.
a) A standard percussion map specifies 47 percussion sounds.
b) Where a note appears on a musical score determines what percussion instrument is being struck: a bongo drum, a cymbal.
c) Other requirements for General MIDI compatibility: MIDI device must
support all 16 channels; a device must be multitimbral (i.e., each
channel can play a different instrument/program); a device must be
polyphonic (i.e., each channel is able to play many voices); and there
must be a minimum of 24 dynamically allocated voices.
General MIDI Level2: An extended general MIDI has recently been defined, with a standard .smf Standard MIDI
File format defined inclusion of extra character information, such as karaoke lyrics.
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MIDI to WAV Conversion

Some programs, such as early versions of Premiere, cannot
include .mid files instead, they insist on .wav format files.
a) Various shareware programs exist for approximating a
reasonable conversion between MIDI and WAV formats.
b) These programs essentially consist of large lookup files
that try to substitute pre-defined or shifted WAV output
for MIDI messages, with inconsistent success.
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6.3 Quantization and Transmission of Audio

Coding of Audio: Quantization and transformation of data
are collectively known as coding of the data.
a) For audio, the -law technique for companding audio signals is usually
combined with an algorithm that exploits the temporal redundancy
present in audio signals.
b) Differences in signals between the present and a past time can reduce
the size of signal values and also concentrate the histogram of pixel
values (differences, now) into a much smaller range.
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c) The result of reducing the variance of values is that lossless compression methods produce a bitstream with shorter bit lengths for more
likely values ( expanded discussion in Chap.7).
In general, producing quantized sampled output for audio

is called PCM (Pulse Code Modulation). The differences
version is called DPCM (and a crude but efficient variant is
called DM). The adaptive version is called ADPCM.
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Pulse Code Modulation

The basic techniques for creating digital signals from analog
signals are sampling and quantization.
Quantization consists of selecting breakpoints in magnitude,
and then re-mapping any value within an interval to one of
the representative output levels. Repeat of Fig. 6.2:
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Amplitude
Amplitude
Time
Time
(a)
(b)
Fig. 6.2: Sampling and Quantization.
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a) The set of interval boundaries are called decision boundaries, and

the representative values are called reconstruction levels.
b) The boundaries for quantizer input intervals that will all be mapped
into the same output level form a coder mapping.
c) The representative values that are the output values from a quantizer
are a decoder mapping.
d) Finally, we may wish to compress the data, by assigning a bit stream
that uses fewer bits for the most prevalent signal values (Chap. 7).
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Every compression scheme has three stages:

A. The input data is transformed to a new representation that is easier
or more efficient to compress.
B. We may introduce loss of information. Quantization is the main lossy
step we use a limited number of reconstruction levels, fewer than
in the original signal.
C. Coding. Assign a codeword (thus forming a binary bitstream) to
each output level or symbol. This could be a fixed-length code, or a
variable length code such as Huffman coding (Chap. 7).
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For audio signals, we first consider PCM for digitization. This

leads to Lossless Predictive Coding as well as the DPCM
scheme; both methods use differential coding. As well, we
look at the adaptive version, ADPCM, which can provide
better compression.
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PCM in Speech Compression

Assuming a bandwidth for speech from about 50 Hz to about
10 kHz, the Nyquist rate would dictate a sampling rate of
20 kHz.
(a) Using uniform quantization without companding, the minimum sample size we could get away with would likely be about 12 bits. Hence
for mono speech transmission the bit-rate would be 240 kbps.
(b) With companding, we can reduce the sample size down to about
8 bits with the same perceived level of quality, and thus reduce the
bit-rate to 160 kbps.
(c) However, the standard approach to telephony in fact assumes that
the highest-frequency audio signal we want to reproduce is only about
4 kHz. Therefore the sampling rate is only 8 kHz, and the companded
bit-rate thus reduces this to 64 kbps.
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However, there are two small wrinkles we must also address:
1. Since only sounds up to 4 kHz are to be considered, all

other frequency content must be noise. Therefore, we
should remove this high-frequency content from the analog input signal. This is done using a band-limiting filter
that blocks out high, as well as very low, frequencies.
Also, once we arrive at a pulse signal, such as that in
Fig. 6.13(a) below, we must still perform DA conversion
and then construct a final output analog signal. But,
effectively, the signal we arrive at is the staircase shown
in Fig. 6.13(b).
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Amplitude
4
3
2
1
0
1
2
3
4
Original signal
t
PCM signals
(a)
Amplitude
Amplitude
4
3
2
1
0
1
2
3
4
(b)
4
3
2
1
0
1
2
3
4
(c)
Fig. 6.13: Pulse Code Modulation (PCM). (a) Original analog signal
and its corresponding PCM signals. (b) Decoded staircase signal. (c)
Reconstructed signal after low-pass filtering.
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2. A discontinuous signal contains not just frequency components due to the original signal, but also a theoretically
infinite set of higher-frequency components:
(a) This result is from the theory of Fourier analysis, in
signal processing.
(b) These higher frequencies are extraneous.
(c) Therefore the output of the digital-to-analog converter
goes to a low-pass filter that allows only frequencies
up to the original maximum to be retained.
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The complete scheme for encoding and decoding telephony

signals is shown as a schematic in Fig. 6.14. As a result
of the low-pass filtering, the output becomes smoothed and
Fig. 6.13(c) above showed this effect.
Input analog
speech signal
-law or
A-law
compressor
Bandlimiting
filter
Linear PCM
Transmission
Output analog
speech signal
-law or
A-law
expander
Low-pass
filter
Digital-to-analog
converter
Fig. 6.14: PCM signal encoding and decoding.

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Differential Coding of Audio

Audio is often stored not in simple PCM but instead in a
form that exploits differences which are generally smaller
numbers, so offer the possibility of using fewer bits to store.
(a) If a time-dependent signal has some consistency over time
(temporal redundancy), the difference signal, subtracting the current sample from the previous one, will have a
more peaked histogram, with a maximum around zero.
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(b) For example, as an extreme case the histogram for a linear

ramp signal that has constant slope is flat, whereas the
histogram for the derivative of the signal (i.e., the differences, from sampling point to sampling point) consists of
a spike at the slope value.
(c) So if we then go on to assign bit-string codewords to
differences, we can assign short codes to prevalent values
and long codewords to rarely occurring ones.
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Lossless Predictive Coding

Predictive coding: simply means transmitting differences
predict the next sample as being equal to the current
sample; send not the sample itself but the difference between
previous and next.
(a) Predictive coding consists of finding differences, and transmitting
these using a PCM system.
(b) Note that differences of integers will be integers. Denote the integer
input signal as the set of values fn . Then we predict values fc
n as
simply the previous value, and define the error en as the difference
between the actual and the predicted signal:
fcn = fn1
en = fn fcn
70
(6.12)
Li & Drew
(c) But it is often the case that some function of a few of the previous values, fn1 , fn2, fn3 , etc., provides a better prediction.
Typically, a linear predictor function is used:
fcn =
2X
to 4
ank fnk
(6.13)
k=1
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The idea of forming differences is to make the histogram of

sample values more peaked.
(a) For example, Fig.6.15(a) plots 1 second of sampled speech at 8 kHz,
with magnitude resolution of 8 bits per sample.
(b) A histogram of these values is actually centered around zero, as in
Fig. 6.15(b).
(c) Fig. 6.15(c) shows the histogram for corresponding speech signal
differences: difference values are much more clustered around zero
than are sample values themselves.
(d) As a result, a method that assigns short codewords to frequently
occurring symbols will assign a short code to zero and do rather
well: such a coding scheme will much more efficiently code sample
differences than samples themselves.
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Count
Magnitude
0.04
0.0
0.04
2000
4000
Samples
6000
8000
1.0
0.5
0.0
0.5
Sample value
1.0
Count
1.0
0.5
0.0
0.5
Sample difference
1.0
Fig. 6.15: Differencing concentrates the histogram. (a): Digital speech

signal. (b): Histogram of digital speech signal values. (c): Histogram of
digital speech signal differences.
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One problem: suppose our integer sample values are in the

range 0..255. Then differences could be as much as -255..255
weve increased our dynamic range (ratio of maximum to
minimum) by a factor of two need more bits to transmit
some differences.
(a) A clever solution for this: define two new codes, denoted SU and SD,
standing for Shift-Up and Shift-Down. Some special code values will
be reserved for these.
(b) Then we can use codewords for only a limited set of signal differences,
say only the range 15..16. Differences which lie in the limited range
can be coded as is, but with the extra two values for SU, SD, a value
outside the range 15..16 can be transmitted as a series of shifts,
followed by a value that is indeed inside the range 15..16.
(c) For example, 100 is transmitted as: SU, SU, SU, 4, where (the codes
for) SU and for 4 are what are transmitted (or stored).
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Lossless predictive coding the decoder produces the same

signals as the original. As a simple example, suppose we
devise a predictor for fcn as follows:
fcn = b 1
2 (fn1 + fn2)c
en = fn fcn
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(6.14)

Li & Drew
Lets consider an explicit example. Suppose we wish to code

the sequence f1, f2, f3, f4, f5 = 21, 22, 27, 25, 22. For the
purposes of the predictor, well invent an extra signal value
f0, equal to f1 = 21, and first transmit this initial value,
uncoded:
fc2 = 21 , e2 = 22 21 = 1;
1 (22 + 21)c = 21,
fc3 = b 1
(f
+
f
)c
=
b
2
1
2
2
e3 = 27 21 = 6;
1 (27 + 22)c = 24,
fc4 = b 1
(f
+
f
)c
=
b
3
2
2
2
e4 = 25 24 = 1;
1 (25 + 27)c = 26,
fc5 = b 1
(f
+
f
)c
=
b
4
3
2
2
e5 = 22 26 = 4
76
(6.15)

Li & Drew
The error does center around zero, we see, and coding (assigning bit-string codewords) will be efficient. Fig. 6.16
shows a typical schematic diagram used to encapsulate this
type of system:
77

Li & Drew
fn
en
Predictor
fn
en
fn Reconstructed
Predictor
fn
Fig. 6.16: Schematic diagram for Predictive Coding encoder

and decoder.
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DPCM
Differential PCM is exactly the same as Predictive Coding,
except that it incorporates a quantizer step.
(a) One scheme for analytically determining the best set of quantizer
steps, for a non-uniform quantizer, is the Lloyd-Max quantizer, which
is based on a least-squares minimization of the error term.
(b) Our nomenclature: signal values: fn the original signal, fbn the
predicted signal, and fen the quantized, reconstructed signal.
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(c) DPCM: form the prediction; form an error en by subtracting the prediction from the actual signal; then quantize the error to a quantized
version, een .
The set of equations that describe DPCM are as follows:
fbn = function of (fen1 , , fen2 , fen3 , ...) ,
en = fn fbn ,
een = Q[en] ,
transmit codeword(een) ,
(6.16)
reconstruct: fen = fbn + een .

Then codewords for quantized error values een are produced using
entropy coding, e.g. Huffman coding (Chapter 7).
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(d) The main effect of the coder-decoder process is to produce reconstructed, quantized signal values fn = fn +
en .
P
2
The distortion is the average squared error [ N

n=1 (fn fn ) ]/N ; one
often plots distortion versus the number of bit-levels used. A LloydMax quantizer will do better (have less distortion) than a uniform
quantizer.
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Li & Drew
For speech, we could modify quantization steps adaptively by estimating

the mean and variance of a patch of signal values, and shifting quantization steps accordingly, for every block of signal values. That is, starting
at time i we could take a block of N values fn and try to minimize the
quantization error:
min
i+N
X1
2
(fn Q[fn])
(6.17)
n=i
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Li & Drew
Since signal differences are very peaked, we could model them using a
Laplacian probability distribution function, which is strongly peaked at
zero: it looks like
2
l(x) = (1/ 2 )exp( 2|x|/)
for variance 2.
So typically one assigns quantization steps for a quantizer with nonuniform steps by assuming signal differences, dn are drawn from such a
distribution and then choosing steps to minimize
min
i+N
X1
2
(dn Q[dn]) l(dn) .
(6.18)
n=i
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Li & Drew
This is a least-squares problem, and can be solved iteratively == the

Lloyd-Max quantizer.
Schematic diagram for DPCM:
fn
~
en
en
Symbol
coder
Quantizer
fn
Symbol
decoder
~
fn
Predictor
~
en
Binary stream
~
fn Reconstructed
+
Predictor
fn
Binary stream
Fig. 6.17: Schematic diagram for DPCM encoder and decoder
84

Li & Drew
Notice that the quantization noise, fn fn , is equal to the quantization

en .
effect on the error term, en
Lets look at actual numbers: Suppose we adopt the particular predictor
below:
fn = trunc

fn1 + fn2 /2
(6.19)
so that en = fn fn is an integer.
As well, use the quantization scheme:
en = Q[en] = 16 trunc ((255 + en) /16) 256 + 8

fn = fn +
en
85
(6.20)

Li & Drew
First, we note that the error is in the range 255..255, i.e., there are
511 possible levels for the error term. The quantizer simply divides the
error range into 32 patches of about 16 levels each. It also makes the
representative reconstructed value for each patch equal to the midway
point for each group of 16 levels.
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Li & Drew
Table 6.7 gives output values for any of the input codes: 4-bit codes are
mapped to 32 reconstruction levels in a staircase fashion.
Table 6.7 DPCM quantizer reconstruction levels.
en in range
-255 .. -240
-239 .. -224
.
.
.
-31 .. -16
-15 .. 0
1 .. 16
17 .. 32
.
.
.
225 .. 240
241 .. 255
Quantized to value
-248
-232
.
.
.
-24
-8
8
24
.
.
.
232
248
87

Li & Drew
As an example stream of signal values, consider the set of values:

f1
130
f2
150
f3
140
f4
200
f5
230 .
Prepend extra values f = 130 to replicate the first value, f1 . Initialize

with quantized error
e1 0, so that the first reconstructed value is exact:
f1 = 130. Then the rest of the values calculated are as follows (with
prepended values in a box):
f
e
e
f
=
=
=
=
130
0
0
130
, 130, 142, 144, 167

,
20, 2,
56,
63
,
24, 8,
56,
56
, 154, 134, 200, 223
On the decoder side, we again assume extra values f equal to the correct
value f1 , so that the first reconstructed value f1 is correct. What is
received is
en, and the reconstructed fn is identical to that on the encoder
side, provided we use exactly the same prediction rule.
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Li & Drew
DM
DM (Delta Modulation): simplified version of DPCM. Often
used as a quick AD converter.
1. Uniform-Delta DM: use only a single quantized error value, either
positive or negative.
(a) a 1-bit coder. Produces coded output that follows the original
signal in a staircase fashion. The set of equations is:
fn = fn1 ,
en = fn fn = fn fn1 ,
en =
+k if en > 0, where k is a constant

k otherwise
(6.21)
fn = fn +
en .
Note that the prediction simply involves a delay.
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Li & Drew
(b) Consider actual numbers: Suppose signal values are

f1
10
f2
11
f3
13
f4
15 .
As well, define an exact reconstructed value fe1 = f1 = 10.

(c) E.g., use step value k = 4:
f2 = 10, e2 = 11 10 = 1,
e2 = 4,
f2 = 10 + 4 = 14
f3 = 14, e3 = 13 14 = 1,
e3 = 4, f3 = 14 4 = 10
f4 = 10, e4 = 15 10 = 5,
e4 = 4,
f4 = 10 + 4 = 14 .
The reconstructed set of values 10, 14, 10, 14 is close to the
correct set 10, 11, 13, 15.
(d) However, DM copes less well with rapidly changing signals. One
approach to mitigating this problem is to simply increase the sampling, perhaps to many times the Nyquist rate.
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Li & Drew
2. Adaptive DM: If the slope of the actual signal curve is high, the
staircase approximation cannot keep up. For a steep curve, should
change the step size k adaptively.
One scheme for analytically determining the best set of quantizer
steps, for a non-uniform quantizer, is Lloyd-Max.
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Li & Drew
ADPCM
ADPCM (Adaptive DPCM) takes the idea of adapting the
coder to suit the input much farther. The two pieces that
make up a DPCM coder: the quantizer and the predictor.
1. In Adaptive DM, adapt the quantizer step size to suit the
input. In DPCM, we can change the step size as well as
decision boundaries, using a non-uniform quantizer.
We can carry this out in two ways:
(a) Forward adaptive quantization: use the properties of the input
signal.
(b) Backward adaptive quantizationor: use the properties of the
quantized output. If quantized errors become too large, we should
change the non-uniform quantizer.
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Li & Drew
2. We can also adapt the predictor, again using forward or

backward adaptation. Making the predictor coefficients
adaptive is called Adaptive Predictive Coding (APC):
(a) Recall that the predictor is usually taken to be a linear function of
previous reconstructed quantized values, fn .
(b) The number of previous values used is called the order of the
predictor. For example, if we use M previous values, we need M
coefficients ai, i = 1..M in a predictor
fn =
M
X
aifni
(6.22)
i=1
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Li & Drew
However we can get into a difficult situation if we try to

change the prediction coefficients, that multiply previous quantized values, because that makes a complicated set of equations to solve for these coefficients:
(a) Suppose we decide to use a least-squares approach to solving a minimization trying to find the best values of the ai:
min
N
X
(fn fn )2
(6.23)
n=1
(b) Here we would sum over a large number of samples fn , for the current
patch of speech, say. But because fn depends on the quantization
we have a difficult problem to solve. As well, we should really be
changing the fineness of the quantization at the same time, to suit
the signals changing nature; this makes things problematical.
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Li & Drew
(c) Instead, one usually resorts to solving the simpler problem that results
from using not fn in the prediction, but instead simply the signal fn
itself. Explicitly writing in terms of the coefficients ai, we wish to
solve:
min
N
X
n=1
(fn
M
X
aifni )2
(6.24)
i=1
Differentiation with respect to each of the ai, and setting to zero,

produces a linear system of M equations that is easy to solve. (The
set of equations is called the Wiener-Hopf equations.)
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Li & Drew
Fig. 6.18 shows a schematic diagram for the ADPCM coder

and decoder:
Encoder
64 kbps A-law
or u-law
PCM input
Convert to
uniform PCM
fn
en
+
-
Adaptive
quantizer
~en
32 kbps
output
+
Adaptive
predictor
^f
n
~f
Decoder
32 kbps
input
.e
^f
n
fn
Convert to
PCM
64 kbps A-law
or u-law
PCM output
Adaptive
predictor
Fig. 6.18: Schematic diagram for ADPCM encoder and decoder
96

Li & Drew
6.4 Further Exploration

Link to Further Exploration for Chapter 6.
The Useful links included are:
An excellent discussion of the use of FM to create synthetic sound.
An extensive list of audio file formats.
CD audio file formats are somewhat different. The main music format
is called red book audio. A good description of various CD formats
is on the website.
A General MIDI Instrument Patch Map, along with a General MIDI
Percussion Key Map.
A link to good tutorial on MIDI and wave table music synthesis.
A link to a java program for decoding MIDI streams.
A good multimedia/sound page, including a source for locating Internet sound/music materials.
97

Li & Drew

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Fundamentals of Multimedia, Chapter 6

6.1 Digitization of Sound

c Prentice Hall 2003

Fundamentals of Multimedia, Chapter 6

6.1 Digitization of Sound

c Prentice Hall 2003

Fundamentals of Multimedia, Chapter 6

(c) Even though such pressure waves are longitudinal, they

c Prentice Hall 2003

Fundamentals of Multimedia, Chapter 6

c Prentice Hall 2003

Fundamentals of Multimedia, Chapter 6

c Prentice Hall 2003

Fundamentals of Multimedia, Chapter 6

The graph in Fig. 6.1 has to be made digital in both time

c Prentice Hall 2003

Fundamentals of Multimedia, Chapter 6

Sampling and Quantization. (a): Sampling the analog signal

in the time dimension. (b): Quantization is sampling the analog signal

c Prentice Hall 2003

Fundamentals of Multimedia, Chapter 6

Thus to decide how to digitize audio data we need to answer

1. What is the sampling rate?

c Prentice Hall 2003

Fundamentals of Multimedia, Chapter 6

c Prentice Hall 2003

Fundamentals of Multimedia, Chapter 6

Fig. 6.3: Building up a complex signal by superposing sinusoids

c Prentice Hall 2003

Fundamentals of Multimedia, Chapter 6

c Prentice Hall 2003

Fundamentals of Multimedia, Chapter 6

The Nyquist theorem states how frequently we must sample

c Prentice Hall 2003

Fundamentals of Multimedia, Chapter 6

c Prentice Hall 2003

Fundamentals of Multimedia, Chapter 6

c Prentice Hall 2003

Fundamentals of Multimedia, Chapter 6

Nyquist Theorem: If a signal is band-limited, i.e., there is

ftrue < fsampling < 2 ftrue

c Prentice Hall 2003

Fundamentals of Multimedia, Chapter 6

Apparent frequency (kHz)

In general, the apparent frequency of a sinusoid is the lowest

Fig. 6.5: Folding of sinusoid frequency which is sampled at

c Prentice Hall 2003

Fundamentals of Multimedia, Chapter 6

Signal to Noise Ratio (SNR)

c Prentice Hall 2003

Fundamentals of Multimedia, Chapter 6

a) The power in a signal is proportional to the square of the

Power 10; Signal Voltage 20.

c Prentice Hall 2003

Fundamentals of Multimedia, Chapter 6

The usual levels of sound we hear around us are described

c Prentice Hall 2003

Fundamentals of Multimedia, Chapter 6

Signal to Quantization Noise Ratio (SQNR)

c Prentice Hall 2003

Fundamentals of Multimedia, Chapter 6

The quality of the quantization is characterized by the Signal

c Prentice Hall 2003